May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Best database setup for high volume
by Andy Ayers 29 May '12

29 May '12

Hi, Can anyone tell me what the best database setup is for dealing with high call volumes? The background: I've been running with the standard SQLite system for about 3 years without issue but recently am getting a lot of database corruption errors ('database disk image is malformed'). Easily solved by deleting the db and restarting but on occasion it brings my switch down. I've tried upgrading to odbc and mysql but hit 2 problems: FLAG_MULTIPLE_STATEMENTS Error in my_thread_global_end() Both of which are mentioned in the user groups but not with any solutions that worked. I've just upgraded to version 1.2 so this may solve the corruption problems but would really like to get my system set up to handle as much traffic as possible. Any advice or suggestions much appreciated. Kind regards Andy

9 28

add international prefix to gateway caller ID
by Sherif Omran 29 May '12

29 May '12

Hello guys, I have a german gateway among other gateways going to the same dialling plan and it does not send the interational caller ID prefix so I want to add 0049 and strip the first 0 in the caller ID sent. But Is it possible to add a condition inside the gateway setting as follows (action applicaiton set ... etc)? <gateway name="trunk_1000"> <param name="realm" value="bluesip.net"/> <param name="username" value="xxxxxxxx"/> <param name="password" value="xxxxxxxx"/> <param name="register" value="true"/> <action application="set" data="effective_caller_id_number=0${caller_id_number}"/> <param name="extension" value="kgvm-1002"/> </gateway I don't want to add it to the dialing plan in order not to affect the caller id from other gateways. Thank you Sherif

2 1

how to run Lua commands in freeswitch?
by Samira Mh 29 May '12

29 May '12

i have created the file named "helloworld.lua" like this: session:answer(); session:streamFile("blah.wav"); session:hangup(); and it working correctly, **now i want to use lua command -not freeswitch command -in lua for example run: op='+' if op == "+" then r = a + b elseif op == "-" then r = a - b elseif op == "*" then r = a*b elseif op == "/" then r = a/b else error("invalid operation") end but how can do that? thanks alot

2 2

Bridge incoming call to external number
by ocset 29 May '12

29 May '12

Hi Setup Windows 7 Freeswitch GXW4104 FXO 2 x POTS lines I have setup Freeswitch to bridge an incoming call to an external mobile phone using a GXW4104 FXO device. The bridge works great but I am having a real headache with the connection not being disconnected when the users hang up. There is no disconnection issue when a call comes in on either line and is connected to an internal extension (like ext 1001). It is when one line is bride to the second that the issue occurs. <extension name="pots_0155555555"> <condition field="destination_number" expression="^(0155555555)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=30"/> <action application="bridge" data="sofia/internal/0766666666@192.168.1.160:5060 <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain} 1001"/> </condition> </extension> 1. Would this be a limitation of the GXW4104 2. Is it possible to write a "lua" script that could manage the disconnect some how? 3. If I enable "Polarity Reversal" under "FXO Termination" then the disconnect works but I have issues with sound and the system not registering when someone picks up. Please help me fix/diagnose this problem. Thanks O

2 2

"bash" programming on freeswitch?
by Samira Mh 29 May '12

29 May '12

how to run bash script programming in freeswitch using AGI (like freeswitch)? i think freeswitch support JAVA/PERL/PYTHON/RUBY programming

3 5

Bobsled
by mazilo 29 May '12

29 May '12

Is is possible to hack http://bobsled.com BobSled so that it will work with FS (like Google Voice)? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Bobsled-tp7579077.html Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

Re: [Freeswitch-users] RTP media issue
by Mr Nathan Downes 28 May '12

28 May '12

I copy /usr/local/freeswitch before making current in case something goes wrong, so I have quite a few backups, I made current and it didnt work so I deleted that folder then copied a backup back. Then though I had an idea why it didnt run (didnt check log straight away) made current again, but forgot a backup. Then when it didnt work and I found that I copied another backup back.. it was from February. Completely my fault and I shouldnt be doing it at midnight when tired J As for the 3 possible things to try to fix it, at 9:23am on Friday the people we trunk with found an issue on their IP network and corrected it, so I have 0% packet loss from the upstream provider again as we both are in the same data centre. Ill keep tracing until I find a call with a missing packet so I can see the effect. I have some captures at work using tshark Ill send across tomorrow so you can check its not staying in early media, I see an ACK after the 100, but not sure that is all good. Pcapsipdump appears to only capture calls in one direction on that side of things as well. Ill sort out the build and get it current, and try and locate a way to simulate that one lost packet from their side, and we will see how those fixes have gone, the call quality has been fine at the site since they resolved that packet loss. From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, 29 May 2012 5:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue Lost changes in what? This is the reason I don't like dealing with issues on the list instead of Jira. You need to be able to run git head to reproduce an issue its our standard policy. You should really examine your environment and find out why you can't it may be a clue, you should download and build a fresh git checkout. On May 28, 2012 8:53 AM, "Mr Nathan Downes" <nathandownes(a)hotmail.com> wrote: Hi, Made current as requested but nothing works afterwards, freeswitch.log has this in it 2012-05-28 23:39:41.704128 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: tls_version** Had to copy older version back, lost a few changes from last month L From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, 27 May 2012 10:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue Also update, you are running git head so you have to update frequently, you rev is a month old. On May 26, 2012 7:43 AM, "Peter Olsson" <peter.olsson(a)visionutveckling.se> wrote: I think the first step is to check the actual signalling. Just as Anthony mentions, the call setup looks weird. Between FS and the ATA (if I understood the different IP's/peers correctly) it seems FS sends early media (183), and it takes the ATA 10 seconds to ACK this. During the time the call is answered on the other end, which dosn't seem to be passed to the ATA (probably because it's so late with the ACK - I'm not sure about that though). So, first check this - it seems to me that FS might still be in "early media" state during the entire call. About the actual RTP, I see one problem here, it's packet 8146 in outbound.pcap. This is a packet that is probably generated by FS (since by that exact time, you're missing one packet from the provider), the problem here is that the timestamp if way off (I'm not sure where FS gets this ts from). It's because of this faulty timestamp that a few packets after this is dropped, since FS won't send packets with a lower timestamp than before. This is also why packet 8152 seems to have a strange timestamp, but it's actually just passed from the other leg, and is the first timestamp that is greater then packet 8146. One possible solution would be to enable rewrite of timestamps (rtp-rewrite-timestamps in the sofia config), also mentioned here: http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio However, I think you also need to check the strangness in the SIP signalling. /Peter ________________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] för Mr Nathan Downes [nathandownes(a)hotmail.com] Skickat: den 26 maj 2012 00:03 Till: 'FreeSWITCH Users Help' Ämne: Re: [Freeswitch-users] RTP media issue Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using <param name="auto-jitterbuffer-msec" value="60"/>, is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added <action application="set" data="sip_jitter_buffer_during_bridge=true" /> because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of <extension name="02-xxxx-9365" > <condition field="context" expression="public"/> <condition field="destination_number" expression="^02xxxx9365;.*$"> <action application="sleep" data="3000"/> <action application="set" data="domain=mgst.xxxxxxxx.com.au"/> <action application="set" data="domain_name=${domain}"/> <action application="set" data="call_direction=inbound"/> <action application="transfer" data="117 XML mgst.xxxxxxxxx.com.au"/> </condition> </extension> It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. <action application="cng_plc"/> <action application="set" data="jitterbuffer_msec=60"/> in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes <nathandownes(a)hotmail.com> wrote: > Hi, > > enable-soa > > <param name="enable-soa" value="true"/> > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message.. Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter(a)gmail.com > To: freeswitch-users(a)lists.freeswitch.org > CC: nathan(a)nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > <param name="enable-soa" value="false"/> > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes <nathandownes(a)hotmail.com> wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound. The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it. In the corresponding outbound trace, between > packet 8144 and 8152, it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152. This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible. This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only. The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 <tel:%2B19193869900> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fbfffef32764666710318! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

1 0

Re: [Freeswitch-users] RTP media issue
by Mr Nathan Downes 28 May '12

28 May '12

Hi, Made current as requested but nothing works afterwards, freeswitch.log has this in it 2012-05-28 23:39:41.704128 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: tls_version** Had to copy older version back, lost a few changes from last month L From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, 27 May 2012 10:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue Also update, you are running git head so you have to update frequently, you rev is a month old. On May 26, 2012 7:43 AM, "Peter Olsson" <peter.olsson(a)visionutveckling.se> wrote: I think the first step is to check the actual signalling. Just as Anthony mentions, the call setup looks weird. Between FS and the ATA (if I understood the different IP's/peers correctly) it seems FS sends early media (183), and it takes the ATA 10 seconds to ACK this. During the time the call is answered on the other end, which dosn't seem to be passed to the ATA (probably because it's so late with the ACK - I'm not sure about that though). So, first check this - it seems to me that FS might still be in "early media" state during the entire call. About the actual RTP, I see one problem here, it's packet 8146 in outbound.pcap. This is a packet that is probably generated by FS (since by that exact time, you're missing one packet from the provider), the problem here is that the timestamp if way off (I'm not sure where FS gets this ts from). It's because of this faulty timestamp that a few packets after this is dropped, since FS won't send packets with a lower timestamp than before. This is also why packet 8152 seems to have a strange timestamp, but it's actually just passed from the other leg, and is the first timestamp that is greater then packet 8146. One possible solution would be to enable rewrite of timestamps (rtp-rewrite-timestamps in the sofia config), also mentioned here: http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio However, I think you also need to check the strangness in the SIP signalling. /Peter ________________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] för Mr Nathan Downes [nathandownes(a)hotmail.com] Skickat: den 26 maj 2012 00:03 Till: 'FreeSWITCH Users Help' Ämne: Re: [Freeswitch-users] RTP media issue Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using <param name="auto-jitterbuffer-msec" value="60"/>, is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added <action application="set" data="sip_jitter_buffer_during_bridge=true" /> because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of <extension name="02-xxxx-9365" > <condition field="context" expression="public"/> <condition field="destination_number" expression="^02xxxx9365;.*$"> <action application="sleep" data="3000"/> <action application="set" data="domain=mgst.xxxxxxxx.com.au"/> <action application="set" data="domain_name=${domain}"/> <action application="set" data="call_direction=inbound"/> <action application="transfer" data="117 XML mgst.xxxxxxxxx.com.au"/> </condition> </extension> It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. <action application="cng_plc"/> <action application="set" data="jitterbuffer_msec=60"/> in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes <nathandownes(a)hotmail.com> wrote: > Hi, > > enable-soa > > <param name="enable-soa" value="true"/> > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message.. Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter(a)gmail.com > To: freeswitch-users(a)lists.freeswitch.org > CC: nathan(a)nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > <param name="enable-soa" value="false"/> > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes <nathandownes(a)hotmail.com> wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound. The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it. In the corresponding outbound trace, between > packet 8144 and 8152, it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152. This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible. This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only. The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 <tel:%2B19193869900> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fbfffef32764666710318! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

3 2

query on freeswitch
by Samira Mh 28 May '12

28 May '12

how to implement mod query odbc on dialplan of freeswitch? the wiki is colud't help me plz help?

2 2

problem with fs_path in mod_callcenter agents
by Jayesh Nambiar 28 May '12

28 May '12

Hello All, I am using mod_callcenter and have the agents definitions as follows: <agents> <agent name="7080(a)test.com" type="callback" contact="[call_timeout=15,sip_h_X-Type=group]sofia/ test.com/7080@test.com;fs_path=$${proxy_ip}" status="Available"/> <agent name="7040(a)test.com" type="callback" contact="[call_timeout=15,sip_h_X-Type=group]sofia/ test.com/7040@test.com;fs_path=$${proxy_ip}" status="Available"/> </agents> I include the fs_path parameter since my agents are not directly registered on freeswitch. The calls need to be sent to a different proxy server where the agents are registered. proxy_ip variable is defined in the vars.xml. Now the problem is as follows: When I start freeswitch, the calls routed to callcenter queues are not sent to the fs_path parameter defined. Instead freeswitch tries to resolve the domain "test.com" and send the call to the resolved IP address. But after I do a "reload -f mod_callcenter", the calls are properly routed to the IP Address mentioned in the fs_path parameter. I am running the 1.2 RC2 version of freeswitch. Can anyone explain why this might happen and why do I exclusively need to reload the callcenter module to make this work. Thanks for any suggestions or pointers to check for this problem. --- Jayesh

1 0